Audio analyzers currently in use here

During tracking, mixing and mixdown I’m utilizing different analyzers whether thats freeware or commercial, hard- or software. Each of them doing a decent job in its very own area:

VU Meter

Always in good use during tracking and mixing mainly for checking channel levels and gainstaging all kinds of plugins. I also love to have a VU right on the mixbus to get a quick visual indication about Peak vs RMS dynamic behaviour.

TBProAudio mvMeter2 is freeware and actually meters not only VU but also RMS, EBU LU as well as PPM. It is also resizeable (VST3 version) and supports different skins.

Spectrum Analyzer I

To me, the Voxengo SPAN is an all-time classic analyzer and ever so reliable. I’ve always used it to have a quick indication about an instruments frequency coverage or the overall frequency balance on the mixbus. There is always one running at the very end of the summing bus in the post-fader section.

Voxengo SPAN is also freeware and highly customizable regarding the analyzer FFT resolution, slope smoothing and ballistics.

Spectrum Analyzer II

Another spectrum analyzer I’m using is Voxengo TEOTE which actually is not only an analyzer but a full spectrum dynamic processor. However, let alone the analyzer itself (fully working in demo mode!) is an excellent assistant when it comes to assess the overall frequency balance. The analyzer does this in regards to a full spectrum noise profile which is adjustable with a Tilt EQ, basically. Very handy for judging deviations (over time) from an ideal frequency response.

Voxengo TEOTE demo version available on their website.

Loudness Metering

I’m leaving all EBU R128 related business to the TC Electronic Clarity M. Since it is a hardware based monitoring solution it always is active here on my desktop no matter what and also serves for double-checking equal RMS levels (for A/B comparisions) and a quick look at the frequency balance from time to time. The hardware is connected via USB (could be SPDIF as well) and is driven by a small remote plugin sitting at the very end of the summing bus in my setup here. It also offers a vector scope and provides audio correlation information. It supports a vast variety of professional metering standards.

Courtesy of Music Tribe IP Ltd.

Image Courtesy of Music Tribe IP Ltd.




effect pedal affairs

Quite recently I had a closer look into the vast amount of (guitar) effect pedals out there. Most are already DSP based which surprised me a little bit since I still ecpected more discrete analog designs after all. While looking for some neat real analog BBD delay I finally stumbled across Fairfield Circuitry’s “Meet Maude” which got me intrigued, having a rather rough look&feel at first sight but some very delicate implementation details under the hood.

Their delay modulation circuit has some randomness build in and also there is a compression circuit in the feedback loop – both designs I’ve also choosen for NastyDLA and which makes a big impact on the overall sound for granted. But the real highlight is the VCF in the delay feedback path which actually appears to be a low-pass gate – a quite unique design and soundwise also different but appealing in its very own regard.

They employed very similar concepts to their vibrato/chorus box “Shallow Water” featuring also random delay modulation and a low pass gate but this time a little bit more prominent on the face plate. On top, their JFET op-amp adds some serious grit to any kind of input signal. All in all, I did not expect such a bold but niche product to exist. If I ever will own such a thingy, there will be a much more detailed review here for sure.

Dynamic 1073/84 EQ curves?

Yes we can! The 1073 and 84 high shelving filters are featuring that classic frequency dip upfront the HF boost itself. Technically speaking they are not shelves but bell curves with a very wide Q but anyway, wouldn’t it be great if that would be program dependent in terms of expanding and compressing according to the curve shape and giving a dynamic frequency response to the program material?

Again, dynamic EQs makes this an easy task today and I just created some presets for the TDR Nova EQ which you can copy right from here (see below after the break). Instructions: Choose one of the 3 presets (one for each specific original frequency setting – 10/12/16kHz) and just tune the Threshold parameter for band IV (dip operation) and band V (boost operation) to fit to the actual mix situation.

They sound pretty much awesome! See also my Nova presets for the mixbus over here and the Pultec ones here.

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Dynamic Pultec EQ curves?

Wouldn’t it be great if the Pultec boost/cut performance would be program dependent? Sort of expanding and compressing according to the boost/cut settings and giving a dynamic frequency response to the program material.

Well, dynamic EQs makes this an easy task today and I just created some presets for the TDR Nova EQ which you can copy right from here (see below after the break). Instructions: Choose one of the 4 presets (one for each specific original frequency setting – 20/30/60/100Hz) and tune the Threshold parameter for band II (boost operation) and band III (cut operation) to fit to the actual mix situation.

See also my presets for the mixbus over here.

[Read more…]

What I like about the Behringer 2600

What I really like about the Behringer 2600 is that it’s not just a plain copy but introduces some real useful improvements over the original concept. Most important to me is the 19″ form factor which not only reduces the originals size and fits in the rack but also remains big enough to enjoy a great user experience while cabling and tweaking things. And they got rid of those speakers! Instead it offers 2 filter revisions to choose from, two of the oscillator sections are now fully featured, the LFO is part of the main chassis now and new additional timing options for the envelopes has been added as well.

On the other hand, the Behringer 2600 sticks to CV gate voltages following original levels which limits full integration in todays modular world quite a bit. However, this is currently not a big deal to me. I only wish they would have made a true analog delay instead of the spring reverb (emulation). Offering audio in, the device also doubles as an excellent analog effect unit which seems to be a little bit underrated in this regard. Given it’s pricepoint, this feature is already something to consider if one is just looking for an outboard analog filter box or an alternative for something like a MS-20.

The ARP 2600 turns 50

And if time allows, watch this awesome documentary about the history and story behind ARP:

Getting the most out of the SPL Tube Vitalizer

In this article I’m going to share some analysis insights but also proposing an easy to follow 3-step approach for finding the sweet spot while processing any kind of material with this device.

Preparing for winter season: room heating with style

So, having now a Tube Vitalizer here on my desk (at least for some time), I was surprised about the lack of usable online reviews and background information. One just finds the usual YT quality stuff which might be entertaining in the best case but also spreads misinformation ever so often. To save those influencers honor it must be said that the Vitalizer concept is really not that easy to grasp and its quirky user experience makes it not easier. The manual itself is a mixed bag since it contains some useful hints and graphs on the one hand but lots of marketing blurb obscuring things on the other. Time to clean up the mess a little bit.

What it actually does

While easily slotted into the “audio exciter” bucket, some more words are needed to describe what it actually does. Technically speaking, the Vitalizer is basically a parallel dynamic equalizer with an actual EQ curve behaviour which aims to mimic equal loudness contours as specified in ISO226. Rather simplified, it can be seen as a high and low frequency shelving EQ to dial in a basic “smile” EQ curve but one which takes hearing related (psychoacoustic) loudness effects into account. It does this also by generating curves differently based on signal levels, hence the term “dynamic EQ”. And wait, it also adds harmonic content galore.

Taming the beast

To obtain an equal loudness contour the main equalizers center frequency must be properly set depending on the tonal balance of the actual source material. This center frequency can be dialed in somewhere between 1k and 20kHz by adjusting the Hi-Mid Freq knob which defines a cross-over point: while frequencies below that point gets attenuated, the higher frequencies gets boosted. However, this attenuation is already a signal level dependent effect. Opposed to that, the LF EQ itself (which actually is not a shelving but a bell type curve) has a fixed frequency tuned to 50Hz and just the desired boost amount needs to be dialed in. The LF curve characteristic can be further altered (Bass soft/tight) which basically thickens or thins out the below 100Hz area. Finally, this EQ path can be compressed now with the Bass Comp option.

A typical EQ curve created by the Vitalizer

On top of the main EQ path, the Tube Vitalizer offers an additional HF boost and compression option which both can be dialed in to complement the LF behaviour in a very similar fashion but in the high frequency department. Internally, both are in a parallel configuration and mixed back into a dry signal path. The according Process Level knob can be seen as a kind of dry/wet option but only for main the EQ part. The upper HF part is mixed back in separately by the Intensity dial.

Gain-Staging is key

For the EQ section as a whole, the Drive knob is the ticket for proper gain-staging. If compression can be dialed in properly for both compressors (as indicated by the blue flashing lights) input gain is in the right ballpark. One might expect to hear actual compression going on but it appears to be a rather gentle leveling effect.

Gain-staging for the output stage has to be concerned separately which might become an issue if the tube stage is activated and operates in shunt limiting mode. Now you have to take care about proper input levels since the Attenuators for both output channels are operating after the limiter and not beforehand.

Tube stage limiting: input (red) vs output (blue)

Which directly leads us to the additional harmonic content created by this device. First of all, there is always additional harmonic content created by this device, no matter what. One might expect the device to not show any such content with the solid state output stage but it actually does. The tube output stage just increases that content but signal level dependent of course and 2nd order harmonics are always part of that content. A serious additional amount of harmonics gets added as soon as the HF filter gets engaged by dialing in Intensity (and LC Filter mode activated!) but this sounds always very smooth and natural in the top end, surprisingly.

Delicious content

Also impressive is the low noisefloor for both output stage modes, tube and solid state. The first one introduces pretty strong channel crosstalk, though.

Workflow – Finding the sweet spot in 3 easy steps

Initial condition:

  • Drive, Bass, Bass Comp and Intensity set to 0
  • Device is properly gain-staged

1. Set Process to 5 and now find the best fit for Hi-Mid Freq for the given source material. For already mixed 2bus stuff you can narrow it down to 2-3kHz most likely.

2. Dial in Bass (either left or right depending on source and taste) and some compression accordingly.

3. Only then dial in some further HF content via Intensity and some compression accordingly. Adjust HF Freq so it basically fits the source/taste.

Workflow – Tweaking just one knob

My good old buddy Bootsy told me this trick which works surprisingly well.

Initial condition:

  • Left most position: Bass
  • Right most position: Bass Comp, High Comp, High Freq
  • 12-o-clock position: Drive, Intensity
  • Hi-Mid-Freq set to 2.5kHz

Now, just dial in some (few) Process Level to taste.

He also recommends to drive the input to some extend (VU hitting the red zone) using the Tube stage in limiter mode while always engaging LC Filter mode for HF.

What loudspeakers and audio transformers do have in common

Or: WTF is “group delay”?

Imagine a group of people visiting an exhibition having a guided tour. One might expect that the group reaches the exhibitions exit as a whole but in reality there might be a part of that group just lagging behind a little bit actually (e.g. just taking their time).

Speaking in terms of frequency response within audio systems now, this sort of delay is refered to as “group delay”, measured in seconds. And if parts of the frequency range do not reach a listeners ear within the very same time this group delay is being refered to as not being constant anymore.

A flat frequency response does not tell anything about this phenomena and group delay must always be measured separately. Just for reference, delays above 1-4ms (depending on the actual frequency) can actually be perceived by human hearing.

This always turned out to be a real issue in loudspeaker design in general because certain audio events can not perceived as a single event in time anymore but are spread across a certain window of time. The root cause for this anomaly typically lies in electrical components like frequency splitters, amplifiers or filter circuits in general but also physical loudspeaker construction patterns like bass reflex ports or transmission line designs.

Especially the latter ones actually do change the group delay for the lower frequency department very prominently which can be seen as a design flaw but on the other hand lots of hifi enthusiast actually do like this low end behaviour which is able to deliver a very round and full bass experience even within a quite small speaker design. In such cases, one can measure more than 20ms group delay within the frequency content below 100Hz and I’ve seen plots from real designs featuring 70ms at 40Hz which is huge.

Such speaker designs should be avoided in mixing or mastering situation where precision and accuracy is required. It’s also one of the reasons why we can still find single driver speaker designs as primary or additional monitoring options in the studios around the world. They have a constant group delay by design and do not mess around with some frequency parts while just leaving some others intact.

As mentioned before, also several analog circuit designs are able to distort the constant group delay and we can see very typical low end group delay shifts within audio transformer coupled circuit designs. Interestingly, even mastering engineers are utilizing such devices – whether to be found in a compressor, EQ or tape machine – in their analog mastering chain.

The renaissance of the Baxandall EQs

Already in 1950, Peter Baxandall designed an analog tone correction circuit which found its way into some million consumer audio devices later on. Today, it is simply referred to as a Baxandall EQ.

What the f*ck is a Baxandall EQ?

Beside its appearance in numerous guitar amplifiers and effects, it made a very prominent reincarnation in the pro audio gear world in 2010 with the Dangerous Music Bax EQ. The concept shines with its very broad curves and gentle slopes which are all about transparancy and so it came to no surprise that this made it into lots of mastering rigs right away.

And it also had a reason that already in 2011 I did an authentic 1:1 emulation of the very same curves within the Baxter EQ plugin but just adding a dual channel M/S layout to better fit the mastering duties. For maximum accuracy and transparancy it already featured oversampling and double-precision filter calculations to that time and it is still one of my personal all time favourite EQs.


During the last 10 years quite a number of devices emerged each showing its very own interpretation of the Baxandall EQ whether thats in hard or software and this was highly anticipated especially in the mastering domain.

A highly deserved revival aka renaissance.

When comparing units be aware that the frequency labeling is not standardized and different frequencies might be declared while giving you same/similar curves. More plots and infos can be found here (german language).

Lets talk about mixing levels (again)

Some years ago we had lots of discussions about proper mixing levels in the digital domain – with mixed (sic!) results, IIRC. Meanwhile, more and more influencers are claiming that targeting -18dBFS with a VU meter readout is the “digital audio sweet spot” and the way forward in terms of plugin gain staging. In practise that would imply mixing digital peak levels at around 0dBFS again but maybe I’ve missed something during my absence in recent years. So, to what mixing levels are you up to in your DAW today?